RadioDroid | radio browser app that uses www | Audio Utils library

 by   segler-alex Java Version: 0.84 License: GPL-3.0

kandi X-RAY | RadioDroid Summary

kandi X-RAY | RadioDroid Summary

RadioDroid is a Java library typically used in Audio, Audio Utils applications. RadioDroid has no bugs, it has no vulnerabilities, it has build file available, it has a Strong Copyleft License and it has low support. You can download it from GitHub.

radio browser app that uses on android

            kandi-support Support

              RadioDroid has a low active ecosystem.
              It has 545 star(s) with 133 fork(s). There are 37 watchers for this library.
              It had no major release in the last 12 months.
              There are 248 open issues and 314 have been closed. On average issues are closed in 21 days. There are 24 open pull requests and 0 closed requests.
              It has a neutral sentiment in the developer community.
              The latest version of RadioDroid is 0.84

            kandi-Quality Quality

              RadioDroid has 0 bugs and 0 code smells.

            kandi-Security Security

              RadioDroid has no vulnerabilities reported, and its dependent libraries have no vulnerabilities reported.
              RadioDroid code analysis shows 0 unresolved vulnerabilities.
              There are 0 security hotspots that need review.

            kandi-License License

              RadioDroid is licensed under the GPL-3.0 License. This license is Strong Copyleft.
              Strong Copyleft licenses enforce sharing, and you can use them when creating open source projects.

            kandi-Reuse Reuse

              RadioDroid releases are available to install and integrate.
              Build file is available. You can build the component from source.

            Top functions reviewed by kandi - BETA

            kandi has reviewed RadioDroid and discovered the below as its top functions. This is intended to give you an instant insight into RadioDroid implemented functionality, and help decide if they suit your requirements.
            • Connects to the media player .
            • Create the menu items menu .
            • Binds the data to an itemViewHolder .
            • start command .
            • Play a station .
            • Creates the preferences which is stored in a resource bundle .
            • On create view .
            • Perform filter filtering
            • Edit or add a server .
            • Update album art art .
            Get all kandi verified functions for this library.

            RadioDroid Key Features

            No Key Features are available at this moment for RadioDroid.

            RadioDroid Examples and Code Snippets

            No Code Snippets are available at this moment for RadioDroid.

            Community Discussions


            How to get a smaller piece of audio from larger audio captured with browser's Web Audio Api
            Asked 2022-Mar-22 at 12:33

            I'm making a speech-to-text tool. I'm capturing audio in real time (using Web audio api from Chrome) and sending it to a server to convert the audio to text.

            I'd like to extract pieces of the whole audio cause I only want to send sentences, avoiding silences. (cause the api I use has a cost). The problem is that I don't know how to convert the whole audio into pieces.

            I was using MediaRecorder to capture the audio



            Answered 2022-Mar-22 at 12:33

            I've found the answer to my own question, I was using the wrong approach.

            What I need to use to get the raw audio inputs and be able to manipulate them is the AudioWorkletProcessor.

            This video helped me to understand the theory behind:


            And this article helped me understand how to make use of it:



            Audio widget within Jupyter notebook is **not** playing. How can I get the widget to play the audio?
            Asked 2022-Mar-15 at 00:07

            I writing my code within a Jupyter notebook in VS Code. I am hoping to play some of the audio within my data set. However, when I execute the cell, the console reports no errors, produces the widget, but the widget displays 0:00 / 0:00 (see below), indicating there is no sound to play.

            Below, I have listed two ways to reproduce the error.

            1. I have acquired data from the hub data store. Looking specifically at the spoken MNIST data set, I cannot get the data from the audio tensor to play


            Answered 2022-Mar-15 at 00:07

            Apologies for the late reply! In the future, please tag the questions with activeloop so it's easier to sort through (or hit us up directly in community slack ->

            Regarding the Free Spoken Digit Dataset, I managed to track the error with your usage of activeloop hub and audio display.

            adding [:,0] to 9th line will help fixing display on Colab as Audio expects one-dimensional data



            FIXED Trying to Pause/Play MP3 file in HTML
            Asked 2022-Mar-10 at 21:03

            Hoping to have two buttons, one to "Pause" one to "Play" the MP3 audio file. Buttons show but do not pause/play the audio, Aswell as the audio does not play whatsoever. Here's my code I'm using. Thanks.




            Answered 2022-Mar-09 at 21:52

            have you tried using javascript for audio? such as



            Download audio file from html with javascript function
            Asked 2022-Feb-24 at 21:25

            I'm trying to download an mp3 file on the click of a button, it downloads, but the file it downloads is not correct, it's a WAY smaller file than the original one (25 bytes), while the original one is 10MB.

            It's my first time working with downloading stuff so I'm pretty clueless.

            Here's some code:

            JS function:



            Answered 2022-Feb-24 at 18:15

            You can try this. Here you have to provide audio file source instead of image source. I did not try this code, But i assure you to it should work!




            Result from audio FFT function makes it near impossible to inspect low/mid frequencies
            Asked 2022-Feb-17 at 17:32

            I am trying to build a graphical audio spectrum analyzer on Linux. I run an FFT function on each buffer of PCM samples/frames fed to the audio hardware so I can see which frequencies are the most prevalent in the audio output. Everything works, except the results from the FFT function only allocate a few array elements (bins) to the lower and mid frequencies. I understand that audio is logarithmic, and the FFT works with linear data. But with so little allocation to low/mid frequencies, I'm not sure how I can separate things cleanly to show the frequency distribution graphically. I have tried with window sizes of 256 up to 1024 bytes, and while the larger windows give more resolution in the low/mid range, it's still not that much. I am also applying a Hann function to each chunk of data to smooth out the window boundaries.

            For example, I test using a mono audio file that plays tones at 120, 440, 1000, 5000, 15000 and 20000 Hz. These should be somewhat evenly distributed throughout the spectrum when interpreting them logarithmically. However, since FFTW works linearly, with a 256 element or 1024 element array only about 10% of the return array actually holds values up to about 5 kHz. The remainder of the array from FFTW contains frequencies above 10-15 kHz.

            Here's roughly the result I'm after:

            But this is what I'm actually getting:

            Again, I understand this is probably working as designed, but I still need a way to get more resolution in the bottom and mids so I can separate the frequencies better.

            What can I do to make this work?



            Answered 2022-Feb-17 at 11:22

            What you are seeing is indeed the expected outcome of an FFT (Fourier Transform). The logarithmic f-axis that you're expecting is achieved by the Constant-Q transform.

            Now, the implementation of the Constant-Q transform is non-trivial. The Fourier Transform has become popular precisely because there is a fast implementation (the FFT). In practice, the constant-Q transform is often implemented by using an FFT, and combining multiple high-frequency bins. This discards resolution in the higher bins; it doesn't give you more resolution in the lower bins.

            To get more frequency resolution in the lower bins of the FFT, just use a longer window. But if you also want to keep the time resolution, you'll have to use a hop size that's smaller than the window size. In other words, your FFT windows will overlap.



            Play multiple tracks
            Asked 2022-Feb-08 at 18:30

            I'm trying to use the tag, and I want to have as many tracks playing as I add. But in the end, the very first track plays in a circle. How can this be fixed?



            Answered 2022-Feb-08 at 12:49
            • Add a skip button
            • when clicked, call a skip function that will
              • pause if there is something playing then
              • clear the playing/paused class clist
              • increment the current to the next track
              • then play



            A problem with sound producing: How to make sound with Fourier coefficients
            Asked 2022-Feb-04 at 23:39

            I'm trying to create a sound using Fourier coefficients.

            First of all please let me show how I got Fourier coefficients.

            (1) I took a snapshot of a waveform from a microphone sound.

            • Getting microphone: getUserMedia()
            • Getting microphone sound: MediaStreamAudioSourceNode
            • Getting waveform data: AnalyserNode.getByteTimeDomainData()

            The data looks like the below: (I stringified Uint8Array, which is the return value of getByteTimeDomainData(), and added length property in order to change this object to Array later)



            Answered 2022-Feb-04 at 23:39

            In golang I have taken an array ARR1 which represents a time series ( could be audio or in my case an image ) where each element of this time domain array is a floating point value which represents the height of the raw audio curve as it wobbles ... I then fed this floating point array into a FFT call which returned a new array ARR2 by definition in the frequency domain where each element of this array is a single complex number where both the real and the imaginary parts are floating points ... when I then fed this array into an inverse FFT call ( IFFT ) it gave back a floating point array ARR3 in the time domain ... to a first approximation ARR3 matched ARR1 ... needless to say if I then took ARR3 and fed it into a FFT call its output ARR4 would match ARR2 ... essentially you have this time_domain_array --> FFT call -> frequency_domain_array --> InverseFFT call -> time_domain_array ... rinse N repeat

            I know Web Audio API has a FFT call ... do not know whether it has an IFFT api call however if no IFFT ( inverse FFT ) you can write your own such function here is how ... iterate across ARR2 and for each element calculate the magnitude of this frequency ( each element of ARR2 represents one frequency and in the literature you will see ARR2 referred to as the frequency bins which simply means each element of the array holds one complex number and as you iterate across the array each successive element represents a distinct frequency starting from element 0 to store frequency 0 and each subsequent array element will represent a frequency defined by adding incr_freq to the frequency of the prior array element )

            Each index of ARR2 represents a frequency where element 0 is the DC bias which is the zero offset bias of your input ARR1 curve if its centered about the zero crossing point this value is zero normally element 0 can be ignored ... the difference in frequency between each element of ARR2 is a constant frequency increment which can be calculated using



            Get Latency of Bluetooth Headphoners UWP C++
            Asked 2022-Jan-18 at 09:08

            I want to make sure the latency between my app and the bluetooth headphones is accounted for, but I have absolutely no idea how I can get this value. The closest thing I found was: BluetoothLEPreferredConnectionParameters.ConnectionLatency which is only available on Windows 11... Otherwise there isn't much to go on.

            Any help would be appreciated.

            Thanks, Peter



            Answered 2022-Jan-18 at 09:08

            It's very difficult to get the exact latency because it is affected by many parameters - but you're on the right track by guessing that the connection parameters are a factor of this equation. I don't have much knowledge on UWP, but I can give you the general parameters that affect the speed/latency, and then you can check their availability in the API or even contact Windows technical team to see if these are supported.

            When you make a connection with a remote device, the following factors impact the speed/latency of the connection:-

            • Connection Interval: this specifies the interval at which the packets are sent during a connection. The lower the value, the higher the speed. The minimum value as per the Bluetooth spec is 7.5ms.
            • Slave Latency: this is the value you originally mentioned - it specifies the number of packets that can be missed before a connection is considered lost. A value of 0 means that you have the fastest most robust connection.
            • Connection PHY: this is the modulation on which the packets are sent. If both devices support 2MPHY, then the connection should be quicker.
            • Data Length/MTU Extension: these are two separate features but I am looping them together becuase the effect is the same - more bytes are sent per packet, which results in a higher throughput. The maximum value is 251 bytes per packet.

            You can find more information about these parameters here:-

            And below are some other links that might help you understand what is supported on UWP:-



            Loading Wave File but there is random nonsense at the end of the data rather than the expected samples
            Asked 2022-Jan-11 at 19:43

            I've got a simple wav header reader i found online a long time ago, i've gotten back round to using it but it seems to replace around 1200 samples towards the end of the data chunk with a single random repeated number, eg -126800. At the end of the sample is expected silence so the number should be zero.

            Here is the simple program:



            Answered 2022-Jan-07 at 21:55

            WAV is just a container for different audio sample formats.

            You're making assumptions on a wav file that would have been OK on Windows 3.11 :) These don't hold in 2021.

            Instead of rolling your own Wav file reader, simply use one of the available libraries. I personally have good experiences using libsndfile, which has been around roughly forever, is very slim, can deal with all prevalent WAV file formats, and with a lot of other file formats as well, unless you disable that.

            This looks like a windows program (one notices by the fact you're using very WIN32API style capital struct names – that's a bit oldschool); so, you can download libsndfile's installer from the github releases and directly use it in your visual studio (another blind guess).



            How do i get the audio frequency from my mic using javascript?
            Asked 2022-Jan-01 at 12:37

            I need to create a sort of like guitar tuner.. thats recognize the sound frequencies and determines in witch chord i am actually playing. Its similar to this guitar tuner that i found online: But i cant figure it out how it works because of the webpack files..I want to make this tool app backendless.. Someone have a clue about how to do this only in the front end?

            i founded some old pieces of code that doesnt work together.. i need fresh ideas



            Answered 2021-Sep-21 at 01:29

            I suppose it'll depend how you're building your application. Hard to help without much detail around specs. Though, here are a few options for you.

            There are a few stream options, for example;

            Or if you're using React;

            Or if you're wanting to go real basic with some vanilla JS;


            Community Discussions, Code Snippets contain sources that include Stack Exchange Network


            No vulnerabilities reported

            Install RadioDroid

            You can download it from GitHub.
            You can use RadioDroid like any standard Java library. Please include the the jar files in your classpath. You can also use any IDE and you can run and debug the RadioDroid component as you would do with any other Java program. Best practice is to use a build tool that supports dependency management such as Maven or Gradle. For Maven installation, please refer For Gradle installation, please refer .


            For any new features, suggestions and bugs create an issue on GitHub. If you have any questions check and ask questions on community page Stack Overflow .
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